FIG. 1 is a block diagram showing a configuration of a circuit of a conventional echo processor. In the diagram, reference number 1 designates a digital to analogue (D/A) converter for converting a received signal in digital form to a sound signal in analogue form, 2 denotes an amplifier for amplifying the sound signal, and 3 indicates a speaker for outputting the acoustic corresponding to the sound signal amplified. Reference number 4 designates a microphone for inputting an acoustic and converting it to a sound signal, 5 denotes an amplifier for amplifying the sound signal, and 6 indicates an analogue to digital (A/D) converter for converting the amplified sound signal to a digital signal. Reference number 7 designates an echo canceller for generating a pseudo echo signal based on the digital received signal, subtracting the pseudo echo signal from the near-end signal converted by and then outputted from the A/D converter 6 in order to eliminate the echo signal component. In the echo canceller 7, reference number 8 designates an adaptive filter for generating the pseudo echo signal based on the received signal, the transmission signal, and a judgment signal. Reference number 9 indicates an adder for adding the near-end signal and the pseudo echo signal. Reference number 10 indicates a double-talk detector for generating the judgment signal for judging whether the received signal is in a silent or near-end signal is in a double-talk state based on the received signal, the near-end signal, and a transmission signal, and outputting the judgment signal to the echo canceller 7.
Next, a description will now be given of the operation of the conventional echo processor.
It is assumed that the received signal in digital form is transmitted from a far-end talker and the sound signal from the microphone 4 contains near-end talker speech and echo. The amplifier 5 amplifies the sound signal from the near-end talker. The A/D converter 6 converts the amplified sound signal to the near-end signal. The echo canceller 7 generates the transmit signal by suppressing the echo signal component from the near-end signal and then outputs it.
The received signal from the far-end talker is inputted to both the echo canceller 7 and the double-talk detector 10, and also converted to an analogue signal, namely the sound signal, by the D/A converter 1. The amplifier 2 amplifies the analogue signal and transmits it to the speaker 3. The speaker 3 receives and outputs it as an acoustic.
The microphone 4 inputs a part of the acoustic outputted from the speaker 3 through the echo path, and mixes the inputted one into the sound signal to be transmitted as the echo signal.
The adaptive filter 8 calculates an adaptive filter coefficient based on the received signal and the transmit signal which is fed back in order to generate the pseudo echo signal.
The adder 9 subtracts the pseudo echo signal from the near-end signal, so that the echo signal component in the near-end signal is suppressedand the transmit signal is thereby generated. In this case, the adaptive filter 8 calculates and updates the adaptive filter coefficient successively in order to adapt the change of the characteristic of the echo path.
The double-talk detector 10 judges whether the near-end signal is in the double-talk state or in the silent state, where in the double-talk state the near-end signal includes both the echo signal and the sound signal from the near-end talker simultaneously, and the received signal is in the silent state. The double-talk detector 10 outputs to the adaptive filter 8 the judgment result as the judgment signal.
In order to prevent any deterioration of the calculation accuracy of the adaptive filter coefficient, the adaptive filter 8 halts the update of the adaptive filter coefficient when it is in the double-talk state or the received signal is in the silent state.
For example, a patent document of Japanese laid-open publication number JP-H10/242891 (Reference 1) has disclosed a technique how to detect the double-talk by the double-talk detector 10.
In the patent document, the detection of the double-talk is performed by the following manner.
The double-talk detector 10 calculates a mean power S of the near-end signal, a mean power X of the received signal, and a means power E of the transmission signal, and detects the double-talk state based on a combination of the following equations (1) to (3):X<p1  (1), S>p2×X(where, p2≦0.5)  (2), andE>p3×S  (3).P1, p2, and p3 in the above equations (1) to (3) are predetermined constants and determined according to an environment.
When the equation (1) is satisfied, the detector 10 judges that the received signal is in the silent state.
In addition, when the equation (1) is not satisfied and the equation (2) is satisfied, the detector 10 judges that the near-end signal is in the double-talk state.
Further, when both the equations (1) and (2) are not satisfied and the equation (3) is satisfied, the detector 10 judges that the near-end signal is in the double-talk state where an amount of the echo suppression is small and the amount of the input sound signal other than the echo signal is large.
Another patent document of Japanese laid-open publication number JP-H9/205388 (Reference 2) has disclosed a conventional echo processor. This echo processor suppresses a component of a low frequency region (hereinafter, referred to as a low frequency component) in a background acoustic inputted through the microphone. In order to achieve this function, the echo processor incorporates a high-pass filter following the A/D converter 6 for the transmit signal, and incorporates an additional high-pass filter, having the same function for cutting a same frequency region, following the input side of the adaptive filter. This configuration enables that both the near-end signal and the received signal to be inputted to the adaptive filter 8 have the same frequency properties and the adaptive filter 8 continues the same calculation accuracy for the adaptive filter coefficient.
Because the conventional echo processor such as the conventional echo processor shown in FIG. 1 and the echo processor disclosed in Reference 2 has the configuration described above, there is a possibility to occur a non-linear distortion because the sound vibration of a high frequency component is prevented by a low frequency component, which cannot be reproduced as an acoustic, inputted to the speaker together with the high frequency component.
In this case, a non-linear distortion occurs in the echo signal inputted through the microphone and the calculation accuracy of the adaptive filter coefficient calculated by the adaptive filter 8 is thereby deteriorated. Further, there is a drawback in which the difference between the pseudo echo signal to be generated and the echo signal becomes large, and the amount of the echo suppression becomes reduced.
In order to eliminate the non-linear distortion occurred in the speaker and to efficiently perform a noise cancelling, a Japanese laid-open publication number JP-H6/202669 (Reference 3) has proposed a conventional technique in which a high-pass filter is inserted at a place before the speaker in order to eliminate the low frequency component in advance which cannot be reproduced as an acoustic.
Although it is possible to apply this technique to the conventional echo processor shown in FIG. 1, where the high-pass filter is inserted between the D/A converter 1 and the amplifier 2, a drawback happens, in which the frequency property of the received signal to be inputted into the adaptive filter 8 is greatly different from that of a low frequency band (hereinafter referred to as a low band) of the echo signal outputted from the speaker 3 and then inputted to the microphone 4. The reason why is that the received signal has a low frequency component, but, the echo signal has not. As a result, the adaptive filter 8 outputs the adaptive filter coefficient with low calculation accuracy, and the difference between the pseudo echo signal to be generated and the echo signal becomes large, so that the amount of the echo suppression is reduced.
We will discuss about the case where the method disclosed in Reference 3 in which the high-pass filter is inserted before the speaker is applied to the echo processor shown in Reference 2. In a case in which a high-pass filter is inserted between the D/A converter and the amplifier at the received signal side, it is difficult to match the frequency components of this high-pass filter and both the high-pass filters placed before the adaptive filter and before the adder because the purpose to suppress the low frequency component is different between those high-pass filters. As a result, the drawback occurs that the characteristic of the low frequency component is greatly different between the received signal to be inputted to the adaptive filter and the echo signal to be inputted to the microphone, this deteriorates the calculation accuracy of the adaptive filter coefficient and the difference between the pseudo echo signal and the echo signal becomes large, and the amount of the echo suppression is thereby reduced.
There is a possibility in which the double-talk detector 10 in the conventional echo processor shown in FIG. 1 takes a wrong judgment for the double-talk in cases where the values “S” and “X” are approximately equal in the equation (2), or the values “S” and “E” are approximately equal in the equation (3). For example, when the value “p2” in the equation (2) is set to a low value so that the double-talk is relatively judged in order to avoid an occurrence of the update for the adaptive filter coefficient with a wrong value, the wrong judgment of the double-talk occurs because the power of the echo signal to be mixed into the near-end signal becomes large (S becomes large) in the case where the distance between the speaker 3 and the microphone 4 is narrow or the amplified values of the amplifiers 2 and 5 are large even if a single-talk (X is large) of only the received signal is used.
When the value “p3” in the equation (3) is set to a low value in order to achieve the same purpose, the wrong judgment of the double-talk also occurs because the calculation accuracy of the adaptive filter coefficient by the adaptive filter 8 becomes reduced even if a single-talk (X is large) of only the received signal is used, and the amount of the echo suppression by the echo canceller 7 becomes less, so that the value E becomes increased towards the value S
Thus, the conventional echo cancellers involve the drawback in which there is a possibility to occur the wrong judgment because it is difficult to distinct the single-talk and the double-talk apparently. As a result, the wrong judgment to halt and start the update of the adaptive filter coefficient occurs, so that the calculation accuracy of the adaptive filter coefficient is deteriorated and the amount of the echo suppression is thereby reduced.
In order to solve the conventional drawbacks described above, an object of the present invention is to provide an echo processor to reduce a non-linier distortion of acoustics outputted from a speaker, to prevent a deterioration of a calculation accuracy of an adaptive filter coefficient in order to eliminate the difference between a pseudo echo signal and an acoustic echo signal, and to prevent the decreasing of the amount of the echo suppression.
In addition, another object of the present invention is to provide an echo processor for judging a double-talk accuracy, to perform to halt and start the update of the adaptive filter coefficient correctly, and to prevent any occurrence of decreasing of an amount of the echo suppression.